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Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp

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Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
This commit is contained in:
Artem Titov
2021-07-28 23:57:33 +02:00
committed by WebRTC LUCI CQ
parent 7f9ab1aa1b
commit 913cfa76ec
72 changed files with 257 additions and 257 deletions
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8
modules/rtp_rtcp/source/rtp_sender_video.h
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@ -67,7 +67,7 @@ class RTPSenderVideo {
Config(const Config&) = delete;
Config(Config&&) = default;
// All members of this struct, with the exception of |field_trials|, are
// All members of this struct, with the exception of `field_trials`, are
// expected to outlive the RTPSenderVideo object they are passed to.
Clock* clock = nullptr;
RTPSender* rtp_sender = nullptr;
@ -91,7 +91,7 @@ class RTPSenderVideo {
// expected_retransmission_time_ms.has_value() -> retransmission allowed.
// `capture_time_ms` and `clock::CurrentTime` should be using the same epoch.
// Calls to this method is assumed to be externally serialized.
// |estimated_capture_clock_offset_ms| is an estimated clock offset between
// `estimated_capture_clock_offset_ms` is an estimated clock offset between
// this sender and the original capturer, for this video packet. See
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time for more
// details. If the sender and the capture has the same clock, it is supposed
@ -208,12 +208,12 @@ class RTPSenderVideo {
RTC_GUARDED_BY(send_checker_);
absl::optional<VideoLayersAllocation> allocation_
RTC_GUARDED_BY(send_checker_);
// Flag indicating if we should send |allocation_|.
// Flag indicating if we should send `allocation_`.
SendVideoLayersAllocation send_allocation_ RTC_GUARDED_BY(send_checker_);
// Current target playout delay.
VideoPlayoutDelay current_playout_delay_ RTC_GUARDED_BY(send_checker_);
// Flag indicating if we need to send |current_playout_delay_| in order
// Flag indicating if we need to send `current_playout_delay_` in order
// to guarantee it gets delivered.
bool playout_delay_pending_;
// Set by the field trial WebRTC-ForceSendPlayoutDelay to override the playout